[asterisk-users] about the Dial application

Hi guys I am working in Kanpur, India. When someone calls to my server i forward the call to someone else by Dial command. After dialing it says Native bridging. And after that I am unable to detect whether the call was answered, the called number was busy or the call was not completed. One more

Re: [asterisk-users] Major problem with 1.4.21 asterisk

Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: > Hopefully the other guy with the problem can test it because this is a > production server and the client is already upset about the p

Re: [asterisk-users] Major problem with 1.4.21 asterisk

Hopefully the other guy with the problem can test it because this is a production server and the client is already upset about the problems this caused for a day or two till I realized what the issue is so I cant risk it. Maybe I can off hours if he cant though. Mike -Original Message

Re: [asterisk-users] Major problem with 1.4.21 asterisk

Yes I forgot to mention, I did need to do kill -9 to finally kill it. We have the exact same bug. Yes mine works for 10 - 20 minutes also. I am glad I am not alone on this. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Wedne

Re: [asterisk-users] Cisco Presence

SIP. Michiel van Baak wrote: > On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote: >> Does anybody have the settings that you use on a Cisco 7970/79x1 to get >> presence? I see the * side settings, but I can't find the Cisco side >> settings anywhere. > > Sip or Skinny ? >> ___

Re: [asterisk-users] Building a Complex IVR

I don't think anyone did, and I was hoping someone would. :) - Original Message From: Steve Murphy <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, June 24, 2008 3:57:48 PM Subject: Re: [asterisk-users] Building a Complex IVR On Mon, 2008-

Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx

On Wednesday 25 June 2008 17:15:26 Stelios Koroneos wrote: > Depending on the gcc version you use you need to set it to produce i486 > code. > The illegal instructions are probably because the default makefile builds > for a later arch. > Also without an fpu and fp kernel emulation don't expect thi

Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx

Depending on the gcc version you use you need to set it to produce i486 code. The illegal instructions are probably because the default makefile builds for a later arch. Also without an fpu and fp kernel emulation don't expect things like dtmf to work. Kernel fp emulation is very slow. Stelios S

Re: [asterisk-users] Google Apps IMAP

Google Apps version might. 2008/6/25 Marc Smith <[EMAIL PROTECTED]>: > Hi, > > Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail > IMAP? If so, does their IMAP implementation support any kind of > "master user" (Dovecot) abililty? Good? Bad? > > --Marc > > _

[asterisk-users] iax2_trunk_queue: Maximum trunk data space exceeded

hi folks. one of the servers i setup recently start exhibiting "iax2_trunk_queue: Maximum trunk data space exceeded" errors. there was only 1 call going on at the time. usually i have to reload chan_iax2.so or restart Asterisk. but the errors came back within a few minutes. i did a google search o

Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!

On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy <[EMAIL PROTECTED]> wrote: > This is just a note that the fixes in the CDRfix4 and CDRfix6 branches > are getting closer to being merged into 1.4, trunk, and 1.6.x. > > If CDR's are important to you, and you ignore this notice, then > you deserve what y

Re: [asterisk-users] SIP vs. SKINNY

On 14:16, Wed 25 Jun 08, Joe Carroll wrote: > Can anyone comment on the performance benefits when comparing sip to skinny ? Most cisco phones work better with the skinny firmware. That is not true when connecting to asterisk though. It all depends on the version of asterisk you are running. I ha

[asterisk-users] SIP vs. SKINNY

Can anyone comment on the performance benefits when comparing sip to skinny ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-use

Re: [asterisk-users] Cisco Presence

On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote: > Does anybody have the settings that you use on a Cisco 7970/79x1 to get > presence? I see the * side settings, but I can't find the Cisco side > settings anywhere. Sip or Skinny ? > > ___ > --

[asterisk-users] Cisco Presence

Does anybody have the settings that you use on a Cisco 7970/79x1 to get presence? I see the * side settings, but I can't find the Cisco side settings anywhere. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 -

[asterisk-users] Res: Asterisk with Nextone using H323

Thank`s all, Chris Ziomkowski wrote: > If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. > Why are you trying to set "softswitch"? I was asked by a costumer, because he could not use a asterisk as a softswitch in the Nextone configuration, so I`m lookin

[asterisk-users] Cisco 7960 Promiscuous Redirect?

List, A Cisco 7960 is registered to servers A and B, where B is the backup server, only used by the 7960 if A is unreachable. That is the behavior of these phones. A call comes from server B to the 7960, which is successful. The 7960 then tries to park the call via an attended transfer, so the

Re: [asterisk-users] play sound on a specific channel

i've seen that there is the PlayDTMF command. Bye On Tue, Jun 24, 2008 at 8:37 AM, nik600 <[EMAIL PROTECTED]> wrote: > any idea? > > On Sat, Jun 14, 2008 at 9:50 AM, nik600 <[EMAIL PROTECTED]> wrote: >> Hi to all >> >> can i play a sound or a dtmf tone on a specific channel using AMI? >> >> Thank

Re: [asterisk-users] included context not being priori tized properly

On Wednesday 25 June 2008 11:39:33 Brian J. Murrell wrote: > On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote: > > That's only true within the same context. ONLY if a match is not found > > in the current context will it go into an included context. > > Ahhh. Well, then that explains it.

[asterisk-users] [Fwd: Bridging an existing PBX in with Asterisk]

Correctionit's a Fujitsu 9600 PBX Have any of you worked with a Fujitsu J5600 PBX before? How about connecting it to a Asterisk server? Currently this customer has two pri's connected to the Fujitsu pbx box. I'll be introducing Asterisk into their environment, and will slowly migrate

[asterisk-users] Bridging an existing PBX in with Asterisk

Have any of you worked with a Fujitsu J5600 PBX before? How about connecting it to a Asterisk server? Currently this customer has two pri's connected to the Fujitsu pbx box. I'll be introducing Asterisk into their environment, and will slowly migrate all users to Asterisk over a period of se

Re: [asterisk-users] Can asterisk support using different ip for rtp?

some vendors(like alcatel-lucent) developed a kind of sip proxy which includes two parts: one sip signaling module and one or more voice modules. voice modules are responsible for receiving/sending voice traffic(RTP). each voice module has its own IP. so , when the sip signaling part sends out "inv

[asterisk-users] Google Apps IMAP

Hi, Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail IMAP? If so, does their IMAP implementation support any kind of "master user" (Dovecot) abililty? Good? Bad? --Marc ___ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] included context not being prioritized properly

On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote: > That's only true within the same context. ONLY if a match is not found in the > current context will it go into an included context. Ahhh. Well, then that explains it. Any thoughts on how to achieve my goal, without having to encode al

Re: [asterisk-users] included context not being prioritized properly

On Wednesday 25 June 2008 10:54:19 Brian J. Murrell wrote: > I have an "outbound-ld" context as follows: > > [ Context 'outbound-ld' created by 'pbx_config' ] > '_1NXXNXX' => 1. Macro(enumdial|${EXTEN}) > [pbx_config] 102. Wait(1) [pbx_config

Re: [asterisk-users] Any SLA alternatives?

> The simplest way to do this is to use a different prefix for dialling out. > For example, if they dial out with a prefix of 9, use the shared number, and > if they dial out with a prefix of 8, use the private number. > > Took the words right out of my mouth. Basically if you have something l

Re: [asterisk-users] Chef-secretary scenario

You have 2 choices to pickup someone's phone with snom's 1: imagine yourself prefix of pickup, let's say 4 exten=>4XX,1,Pickup([EMAIL PROTECTED]) so if u call 4 + phone number you will pickup that one. Second you can add pickupgroup=number for each phone you want to be in the group, and add a dt

[asterisk-users] included context not being prioritized properly

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LIN

Re: [asterisk-users] Any SLA alternatives?

On Wednesday 25 June 2008 10:20:14 Scott Moseman wrote: > I have a group of people who have distinct phone numbers plus a shared > number. The shared number is actually a group that rings through to > all of their direct numbers. I want them to: 1) be able to make > outgoing calls as the shared n

Re: [asterisk-users] Chef-secretary scenario

Though I wonder... The scenario is as follows: I have 4 phones with the following extensions: 11 (SIP/11) 12 (SIP/12) 13 (SIP/13) 15 (SIP/15) Whenever SIP/11 receives a call, it hints the other phones. Is it possible to pick up that call from one of them? The relevant part of my extensions.conf

Re: [asterisk-users] Number portability in other parts of the world.

On Wed, Jun 25, 2008 at 5:28 PM, Dean Collins <[EMAIL PROTECTED]> wrote: > Number portability exists in Australia but mobile numbers only across mobile > carriers and 'pstn' numbers only across pstn carriers. Incidentally, in France my Internet provider (Neuf) will pay for your cell number to be p

Re: [asterisk-users] Number portability in other parts of the world.

On Wed, Jun 25, 2008 at 4:49 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote: > How does a person in Europe go fully VoIP and still keep the main number? > Do they use call forwarding? I shopuld have mentioned too, that if you run your own asterisk you can of course just be connected to your number(

Re: [asterisk-users] Chef-secretary scenario

Thankyou all! I think I've got it working :-D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Number portability in other parts of the world.

On Wed, Jun 25, 2008 at 4:49 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote: > Are phone numbers portable in other countries? Recent legislation in France (and perhaps this means the EU?) have made it so. > Are the same rules and conditions that exist here in the States mirrored > elsewhere? > > >

Re: [asterisk-users] Number portability in other parts of the world.

Number portability exists in Australia but mobile numbers only across mobile carriers and 'pstn' numbers only across pstn carriers. Some of the earlier Voip providers (yeh I'm talking about you Faktortel ) don't allow portabili

[asterisk-users] Any SLA alternatives?

I have a group of people who have distinct phone numbers plus a shared number. The shared number is actually a group that rings through to all of their direct numbers. I want them to: 1) be able to make outgoing calls as the shared number and 2) be able to make outgoing calls as their direct numb

Re: [asterisk-users] Number portability in other parts of the world.

On Wed, Jun 25, 2008 at 10:49:18AM -0400, Alexander Lopez wrote: >Are phone numbers portable in other countries? Depends what country >Are the same rules and conditions that exist here in the States >mirrored elsewhere? >How does a person in Europe go fully VoIP and still keep th

[asterisk-users] Number portability in other parts of the world.

Are phone numbers portable in other countries? Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? Do they use call forwarding? Is their another way to use an origination carr

Re: [asterisk-users] Chef-secretary scenario

Vazquez David wrote: > Yes, I'm using 1.4. And I don't really use sip.conf, but have all my > phones on users.conf. Should I put limitonpeers and call-limit on the > general section of sip.conf? or on each entry in users.conf [general] should be sufficient, so long as having them set as default fo

Re: [asterisk-users] Major problem with 1.4.21 asterisk

On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote: > Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having > major iax2 problems. All of a sudden calls wouldnt come in on the iax2 > DID, and we couldnt make calls out even though everything looked ok. > Also there was usua

Re: [asterisk-users] asterisk seg fault

On Wednesday 25 June 2008 07:42:21 Jerry Geis wrote: > I am running asterisk from svn check out from yesterday Jun 24. > I started with 1.4.20, then 1.4.21 then svn. > > I am getting: > pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed > segment fault. > > I am running debia

Re: [asterisk-users] asterisk seg fault

I managed to catch the whole trace from the seg fault. What is my next step? Program received signal SIGABRT, Aborted. [Switching to Thread -1224033360 (LWP 2440)] 0xb7d77947 in raise () from /lib/tls/libc.so.6 (gdb) where #0 0xb7d77947 in raise () from /lib/tls/libc.so.6 #1 0xb7d790c9 in abort

Re: [asterisk-users] Chef-secretary scenario

Rob Hillis wrote: > Can I assume you're using Asterisk 1.4 and that you've configured your > phones as peers? > > If this is the case, then you need to set limitonpeers to yes and > call-limit to some value in sip.conf. Once this has been done, you > should find that BLF behaves as you expect.

Re: [asterisk-users] Chef-secretary scenario

Can I assume you're using Asterisk 1.4 and that you've configured your phones as peers? If this is the case, then you need to set limitonpeers to yes and call-limit to some value in sip.conf. Once this has been done, you should find that BLF behaves as you expect. Vazquez David wrote: > Hi a

Re: [asterisk-users] Chef-secretary scenario

Grygoriy Dobrovolskyy wrote: > ok, so do you configured you snom phone tu subscribe to monito > extension in asterisk ? > > It is simple to verify: > > core show hints > > Snom's behave exactly like you want when blf enabled. > > 2008/6/25 Vazquez David <[EMAIL PROTECTED] >

Re: [asterisk-users] Chef-secretary scenario

Dr. Michael J. Chudobiak wrote: >> I use Snoms. I know there's the feature. I just don't know how to use >> it, and there's so little documentation on the web.. Anyway, with "see" >> I meant that the "secretary"'s phone would have one of the function keys >> "on" whenever the "chef" is on the phone

[asterisk-users] asterisk seg fault

I am running asterisk from svn check out from yesterday Jun 24. I started with 1.4.20, then 1.4.21 then svn. I am getting: pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed segment fault. I am running debian i386, on a 486 sx machine. I am connecting to the Console/DSP and

Re: [asterisk-users] Chef-secretary scenario

> I use Snoms. I know there's the feature. I just don't know how to use > it, and there's so little documentation on the web.. Anyway, with "see" > I meant that the "secretary"'s phone would have one of the function keys > "on" whenever the "chef" is on the phone (also when he picks it up, > right

Re: [asterisk-users] Chef-secretary scenario

ok, so do you configured you snom phone tu subscribe to monito extension in asterisk ? It is simple to verify: core show hints Snom's behave exactly like you want when blf enabled. 2008/6/25 Vazquez David <[EMAIL PROTECTED]>: > Gordon Henderson wrote: > > On Tue, 24 Jun 2008, Vazquez David wro

Re: [asterisk-users] Centile ipbx, anyone heard of this?

Dont worry i did a lot worse. 2008/6/25 randulo <[EMAIL PROTECTED]>: > Nothing that embarrassing,just didn't want to mention the even more OT > stuff. Everyone already knows I do not too bright things like turning > a phone off and then complaining it doesn't work :) > > On Wed, Jun 25, 2008 at 1

Re: [asterisk-users] Chef-secretary scenario

Gordon Henderson wrote: > On Tue, 24 Jun 2008, Vazquez David wrote: > > >> Hi all, >> >> I'm trying to implement such a scenario where the "Chef" picks up his >> phone and his "secretary" can see that he is busy. Something like blf, I >> guess. >> > > It's like BLF because that's exactly wh

Re: [asterisk-users] Chef-secretary scenario

Grygoriy Dobrovolskyy wrote: > To see? how? what phone do you use? > > Snoms imprement that, you got BLINKING and ON state > BLINKING=calling or being called > ON=on the phone > 2008/6/24 Vazquez David <[EMAIL PROTECTED] > >: > > Hi all, > > I'm trying to implement

Re: [asterisk-users] Centile ipbx, anyone heard of this?

Nothing that embarrassing,just didn't want to mention the even more OT stuff. Everyone already knows I do not too bright things like turning a phone off and then complaining it doesn't work :) On Wed, Jun 25, 2008 at 12:39 PM, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote: > Private messagind:)

Re: [asterisk-users] Centile ipbx, anyone heard of this?

Private messagind:) 2008/6/25 randulo <[EMAIL PROTECTED]>: > On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > > Randy, > > > > This is exactly what was happening when I used an Aastra 480i CT with > > OnSIP. According to OnSIP it's not a supported phone, although the >

Re: [asterisk-users] Can asterisk support using different ip for rtp?

25 jun 2008 kl. 11.15 skrev Raj Jain: > On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <[EMAIL PROTECTED]> wrote: > Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows > RTP to use different IP as SIP ip. > > Is there any way to configure it? GUI or CLI? or , will we support > it in

Re: [asterisk-users] Major problem with 1.4.21 asterisk

Thomas Kenyon wrote: > Michael J. Liberatore wrote: >> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having >> major iax2 problems. All of a sudden calls wouldnt come in on the iax2 >> DID, and we couldnt make calls out even though everything looked ok. >> Also there was usuall

Re: [asterisk-users] Major problem with 1.4.21 asterisk

Michael J. Liberatore wrote: > Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having > major iax2 problems. All of a sudden calls wouldnt come in on the iax2 > DID, and we couldnt make calls out even though everything looked ok. > Also there was usually a hung iax2 channel when

[asterisk-users] misdn issues

Hi, I'm having issues where people call one of my ISDN numbers, and sometime, when we answer, the call is dead. Its only sometimes happens, like one in 15 calls. I have analogue phones pluged into a mediatrix and a couple of pap's, and these connect to the asterisk server. Please could someone

Re: [asterisk-users] Can asterisk support using different ip for rtp?

On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <[EMAIL PROTECTED]> wrote: > Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows > RTP to use different IP as SIP ip. > > Is there any way to configure it? GUI or CLI? or , will we support it in > future? > SIP is decoupled from RTP, so th

Re: [asterisk-users] Centile ipbx, anyone heard of this?

On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > This is very similar to another idea that I once had but never actually > implemented. That is, using a small embedded Asterisk device as a > SIP<>IAX2 protocol translator to facilitate complex NAT traversal. I > thought t

Re: [asterisk-users] Centile ipbx, anyone heard of this?

On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > Randy, > > This is exactly what was happening when I used an Aastra 480i CT with > OnSIP. According to OnSIP it's not a supported phone, although the > newer 57i CT does work with OnSIP. > > It seemed that the phone was lo

Re: [asterisk-users] Can asterisk support using different ip for rtp?

25 jun 2008 kl. 03.26 skrev Jun Yin: > Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows > RTP to use different IP as SIP ip. > > Is there any way to configure it? GUI or CLI? or , will we support > it in future? There's currently no support for that in Asterisk. /O __

Re: [asterisk-users] AS5400 E1 SS7

Alex Balashov wrote: > Nhadie Ramos wrote: > >> Would just like to inquire if anyone here has a setup of asterisk to >> send traffic to AS5400 connected to an SS7-PRI. this is more of a >> AS54 question, as i've been reading and i always stumble upon PGW2200 >> as a requirement to handle SS7

Re: [asterisk-users] AS5400 E1 SS7

Nhadie Ramos wrote: > Would just like to inquire if anyone here has a setup of asterisk to > send traffic to AS5400 connected to an SS7-PRI. this is more of a AS54 > question, as i've been reading and i always stumble upon PGW2200 as a > requirement to handle SS7 signaling on the AS54. Has any

Re: [asterisk-users] Calls drop + "Didn't get a frame from channel" log message

Hi Jean, I have an Asterisk 1.12.18 with about 30 pc each with a Doro SIP phone on an unknown LAN. I think google is useless in cases like this. Many of the system we are working on are in production and we cannot make tests with them so the only hope is to gather infos from people experiencing

[asterisk-users] AS5400 E1 SS7

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.  this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls f

Re: [asterisk-users] mpg123 problem

On Tue, Jun 24, 2008 at 10:06:00PM +0200, Stefan Tichy wrote: > On Sun, Jun 22, 2008 at 12:24:22AM -0700, fateme fatah wrote: > > I want to install mpg123-0.59r on my asterisk server.I downloaded it in > > /usr/src then untared it and I typed these command : > > Just have a look at www.mpg123.org

[asterisk-users] unable to send a fax to a given FAX number

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did s